1. Field of the Invention
This invention relates generally to digital cordless telephones. More particularly, it relates to improved techniques and apparatus for communication of voice data between a base and a remote handset of a digital cordless telephone.
2. Background of Related Art
Digital cordless telephones are popular consumer devices which allow a user in a home or office the freedom to stray hundreds of feet from a base unit. Initially, remote handsets of cordless telephones communicated with their base unit using analog signals. In more recent years, advancements have been made with respect to cordless telephones allowing digital communications between the remote handset and its base unit. The entry of cordless telephones into digital communications generally provides better voice quality because of increased noise rejection, and a somewhat higher range.
FIG. 3 shows relevant features of a conventional digital remote handset of a digital cordless telephone. In FIG. 3, the remote handset of a digital cordless telephone includes a processor (e.g., a digital signal processor (DSP), microprocessor, or microcontroller) 550 comprising a transmitter baseband processor portion 500 and a receiver baseband processor portion 600.
In the transmit direction, a microphone 590 outputs an analog signal to a COder/DECoder (CODEC) 580, which converts the microphone input signal to a digital microphone signal.
The digital microphone signal is encoded into a compressed digital signal and processed by the transmitter baseband processor 500. The compressed digital signal is transmitted by a radio frequency (RF) transmitter 570 and an antenna to a complementary base unit 561.
In the receive direction, an antenna and RF receiver 572 receives a digital signal from the complementary base unit 561. The RF receiver 572 passes the digital signal to a receiver baseband processor portion 600 and decoder 581 for decoding. The decoded digital signal is converted back to analog for output by a speaker 592.
The codec 580 includes an appropriate encoding scheme for use in one direction of a full-duplex audio link, as well as an appropriate decoding scheme for use in the opposite direction of the full-duplex audio link. The encoding scheme utilizes the same algorithm as the decoding scheme, just in an opposite direction. For instance, the codec 580 may provide 32 kb/s adaptive differential pulse code modulation (ADPCM) encoding and decoding. Or, a codec 580 may be selected which provides 64 kb/s μlaw encoding and decoding, or 3.6 kb/s CELP encoding and decoding, or 2.4 kb/s CELP encoding and decoding, etc.
FIG. 4 shows the codec 580 in more detail.
In particular, as shown in FIG. 4, each codec 580 (both in the remote handset 560 as well as in the base unit 561) includes an analog-to-digital (A/D) converter 582 and encoder 300 in a first audio path direction, and a decoder 400 and digital-to-analog (D/A) converter 583 in an opposite audio path direction. The audio paths may be output separately, or sequentially (e.g., using time slots) through an appropriate I/O device 307.
Generally speaking, the encoder 300 is driven by an appropriate encoding algorithm, e.g., a CELP module 309a, and the decoder 400 is similarly driven by a corresponding decoding algorithm, e.g., CELP module 309b. 
Thus, the base unit of a digital cordless telephone system contains encoding and decoding circuitry which is complementary to that contained in a matching remote handset. In either or both the remote handset and/or its base unit, the transmitter and receiver baseband processor portions may be comprised within the same processor, e.g., the same DSP.
The particular encoding and decoding techniques used by encoder 300 and decoder 400 are typically toll-quality. For instance, either 8-bit linear pulse code modulation (PCM), μ-law, A-law, or adaptive differential pulse code modulation (ADPCM) are often used to compress digital speech messages transmitted in both directions between a base and its handset (i.e., from the base unit to its remote handset, and from the remote handset to its base unit).
One commonly used speech encoding/compression algorithm is code excited linear predictive (CELP) based coding. CELP-based algorithms reconstruct speech signals based on a digital model of the human vocal tract. They provide frames of an encoded, compressed bit stream and include short-term spectral linear predictor coefficients, voicing information and gain information (frame and sub frame-based) reconstructable based on a model of the human vocal tract. Whether speech compression can or should be employed often depends on the desired quality of the speech upon reproduction, the sampling rate of the real-time speech, and the available processing capacity to handle speech compression and other associated tasks on-the-fly before storage to voice message memory. CELP bit rates vary, e.g., up to 6.8 Kb/s or more.
In designing a digital cordless telephone, one particular type of encoding and decoding at one particular encoding/decoding bit rate is chosen and fixed by design into the relevant codec block of both the remote handset 560 and the base unit 561.
FIG. 5 shows a real-time speech signal 402 with respect to a noise level 400 determined by a conventional, real-time, time domain-based speech analysis. The noise level 400 represents the maximum desired level of background noise or other unwanted information in speech signal 402. Noise in the real-time speech signal 402 can be a result of many causes, such as audible background noise digitized with the original speech signal at a remote handset, and/or digitization noise caused by the encoding and decoding process itself.
Real-time speech is input to speech encoder 302 for compression into CELP frames. Typical digital encoding techniques, e.g., ADPCM or CELP, raise the noise floor of the digitized speech signal, which either reduces overall sound quality, or demands high performance codec technology for use in both directions to ensure good sound quality. Conventional techniques may use of a dual-direction codec, i.e., COder and DECoder (CODEC), or coding and decoding may be implemented in an appropriate software module processing digitized samples of a voice data stream. In any event, conventional techniques and apparatus utilize the same speech grade coding/decoding in both directions of the full duplex communications between a base unit and its remote handset of a conventional digital cordless telephone.
Accordingly, there is a need for an improved technique and apparatus which avoids the wasted additional cost of providing a higher performance coder/decoder in one direction of a voice communication path of a digital cordless telephone, particularly when only one direction has degraded the overall performance of a full duplex voice communication link.